Initial commit of the atrac1 decoder, not hooked up yet

Originally committed as revision 19811 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 402 insertions, 0 deletions

diff --git a/libavcodec/atrac1.c b/libavcodec/atrac1.c
new file mode 100644
index 0000000000..6d6abe6a4f
--- /dev/null
+++ b/libavcodec/atrac1.c
@@ -0,0 +1,402 @@
+/*
+ * Atrac 1 compatible decoder
+ * Copyright (c) 2009 Maxim Poliakovski
+ * Copyright (c) 2009 Benjamin Larsson
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file libavcodec/atrac1.c
+ * Atrac 1 compatible decoder.
+ * This decoder handles raw ATRAC1 data.
+ */
+
+/* Many thanks to Tim Craig for all the help! */
+
+#include <math.h>
+#include <stddef.h>
+#include <stdio.h>
+
+#include "avcodec.h"
+#include "get_bits.h"
+#include "dsputil.h"
+
+#include "atrac.h"
+#include "atrac1data.h"
+
+#define AT1_MAX_BFU      52                 ///< max number of block floating units in a sound unit
+#define AT1_SU_SIZE      212                ///< number of bytes in a sound unit
+#define AT1_SU_SAMPLES   512                ///< number of samples in a sound unit
+#define AT1_FRAME_SIZE   AT1_SU_SIZE * 2
+#define AT1_SU_MAX_BITS  AT1_SU_SIZE * 8
+#define AT1_MAX_CHANNELS 2
+
+#define AT1_QMF_BANDS    3
+#define IDX_LOW_BAND     0
+#define IDX_MID_BAND     1
+#define IDX_HIGH_BAND    2
+
+/**
+ * Sound unit struct, one unit is used per channel
+ */
+typedef struct {
+    int                 log2_block_count[AT1_QMF_BANDS];    ///< log2 number of blocks in a band
+    int                 num_bfus;                           ///< number of Block Floating Units
+    int                 idwls[AT1_MAX_BFU];                 ///< the word length indexes for each BFU
+    int                 idsfs[AT1_MAX_BFU];                 ///< the scalefactor indexes for each BFU
+    float*              spectrum[2];
+    DECLARE_ALIGNED_16(float,spec1[AT1_SU_SAMPLES]);        ///< mdct buffer
+    DECLARE_ALIGNED_16(float,spec2[AT1_SU_SAMPLES]);        ///< mdct buffer
+    DECLARE_ALIGNED_16(float,fst_qmf_delay[46]);            ///< delay line for the 1st stacked QMF filter
+    DECLARE_ALIGNED_16(float,snd_qmf_delay[46]);            ///< delay line for the 2nd stacked QMF filter
+    DECLARE_ALIGNED_16(float,last_qmf_delay[256+23]);       ///< delay line for the last stacked QMF filter
+} AT1SUCtx;
+
+/**
+ * The atrac1 context, holds all needed parameters for decoding
+ */
+typedef struct {
+    AT1SUCtx            SUs[AT1_MAX_CHANNELS];              ///< channel sound unit
+    DECLARE_ALIGNED_16(float,spec[AT1_SU_SAMPLES]);         ///< the mdct spectrum buffer
+    DECLARE_ALIGNED_16(float,short_buf[64]);                ///< buffer for the short mode
+    DECLARE_ALIGNED_16(float, low[256]);
+    DECLARE_ALIGNED_16(float, mid[256]);
+    DECLARE_ALIGNED_16(float,high[512]);
+    float*              bands[3];
+    float               out_samples[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
+    MDCTContext         mdct_ctx[3];
+    int                 channels;
+    DSPContext          dsp;
+} AT1Ctx;
+
+static float *short_window;
+static float *mid_window;
+DECLARE_ALIGNED_16(static float, long_window[256]);
+static float *window_per_band[3];
+
+/** size of the transform in samples in the long mode for each QMF band */
+static const uint16_t samples_per_band[3] = {128, 128, 256};
+static const uint8_t   mdct_long_nbits[3] = {7, 7, 8};
+
+
+static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, int rev_spec)
+{
+    MDCTContext* mdct_context;
+    int transf_size = 1 << nbits;
+
+    mdct_context = &q->mdct_ctx[nbits - 5 - (nbits>6)];
+
+    if (rev_spec) {
+        int i;
+        for (i=0 ; i<transf_size/2 ; i++)
+            FFSWAP(float, spec[i], spec[transf_size-1-i]);
+    }
+    ff_imdct_half(mdct_context,out,spec);
+}
+
+
+static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
+{
+    int             band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
+    unsigned int    start_pos, ref_pos=0, pos = 0;
+
+    for (band_num=0 ; band_num<AT1_QMF_BANDS ; band_num++) {
+        band_samples = samples_per_band[band_num];
+        log2_block_count = su->log2_block_count[band_num];
+
+        /* number of mdct blocks in the current QMF band: 1 - for long mode */
+        /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
+        num_blocks = 1 << log2_block_count;
+
+        /* mdct block size in samples: 128 (long mode, low & mid bands), */
+        /* 256 (long mode, high band) and 32 (short mode, all bands) */
+        block_size = band_samples >> log2_block_count;
+
+        /* calc transform size in bits according to the block_size_mode */
+        nbits = mdct_long_nbits[band_num] - log2_block_count;
+
+        if (nbits!=5 && nbits!=7 && nbits!=8)
+            return -1;
+
+        if (num_blocks == 1) {
+            at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos], nbits, band_num);
+            pos += block_size; // move to the next mdct block in the spectrum
+        } else {
+            /* calc start position for the 1st short block: 96(128) or 112(256) */
+            start_pos = (band_samples * (num_blocks - 1)) >> (log2_block_count + 1);
+            memset(&su->spectrum[0][ref_pos], 0, sizeof(float) * (band_samples * 2));
+
+            for (; num_blocks!=0 ; num_blocks--) {
+                /* use hardcoded nbits for the short mode */
+                at1_imdct(q, &q->spec[pos], q->short_buf, 5, band_num);
+
+                /* overlap and window between short blocks */
+                q->dsp.vector_fmul_window(&su->spectrum[0][ref_pos+start_pos],
+                    &su->spectrum[0][ref_pos+start_pos],q->short_buf,short_window, 0, 16);
+                start_pos += 32; // use hardcoded block_size
+                pos += 32;
+            }
+        }
+
+        /* overlap and window with the previous frame and output the result */
+        q->dsp.vector_fmul_window(q->bands[band_num], &su->spectrum[1][ref_pos+band_samples/2],
+            &su->spectrum[0][ref_pos], window_per_band[band_num], 0, band_samples/2);
+
+        ref_pos += band_samples;
+    }
+
+    /* Swap buffers so the mdct overlap works */
+    FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
+
+    return 0;
+}
+
+
+static int at1_parse_block_size_mode(GetBitContext* gb, int log2_block_count[AT1_QMF_BANDS])
+{
+    int log2_block_count_tmp, i;
+
+    for(i=0 ; i<2 ; i++) {
+        /* low and mid band */
+        log2_block_count_tmp = get_bits(gb, 2);
+        if (log2_block_count_tmp & 1)
+            return -1;
+        log2_block_count[i] = 2 - log2_block_count_tmp;
+    }
+
+    /* high band */
+    log2_block_count_tmp = get_bits(gb, 2);
+    if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
+        return -1;
+    log2_block_count[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
+
+    skip_bits(gb, 2);
+    return 0;
+}
+
+
+static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, float spec[AT1_SU_SAMPLES])
+{
+    int bits_used, band_num, bfu_num, i;
+
+    /* parse the info byte (2nd byte) telling how much BFUs were coded */
+    su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
+
+    /* calc number of consumed bits:
+        num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
+        + info_byte_copy(8bits) + log2_block_count_copy(8bits) */
+    bits_used = su->num_bfus * 10 + 32 +
+                bfu_amount_tab2[get_bits(gb, 2)] +
+                (bfu_amount_tab3[get_bits(gb, 3)] << 1);
+
+    /* get word length index (idwl) for each BFU */
+    for (i=0 ; i<su->num_bfus ; i++)
+        su->idwls[i] = get_bits(gb, 4);
+
+    /* get scalefactor index (idsf) for each BFU */
+    for (i=0 ; i<su->num_bfus ; i++)
+        su->idsfs[i] = get_bits(gb, 6);
+
+    /* zero idwl/idsf for empty BFUs */
+    for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
+        su->idwls[i] = su->idsfs[i] = 0;
+
+    /* read in the spectral data and reconstruct MDCT spectrum of this channel */
+    for (band_num=0 ; band_num<AT1_QMF_BANDS ; band_num++) {
+        for (bfu_num=bfu_bands_t[band_num] ; bfu_num<bfu_bands_t[band_num+1] ; bfu_num++) {
+            int pos;
+
+            int num_specs = specs_per_bfu[bfu_num];
+            int word_len  = !!su->idwls[bfu_num] + su->idwls[bfu_num];
+            float scale_factor = sf_table[su->idsfs[bfu_num]];
+            bits_used    += word_len * num_specs; /* add number of bits consumed by current BFU */
+
+            /* check for bitstream overflow */
+            if (bits_used > AT1_SU_MAX_BITS)
+                return -1;
+
+            /* get the position of the 1st spec according to the block size mode */
+            pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
+
+            if (word_len) {
+                float   max_quant = 1.0/(float)((1 << (word_len - 1)) - 1);
+
+                for (i=0 ; i<num_specs ; i++) {
+                    /* read in a quantized spec and convert it to
+                     * signed int and then inverse quantization
+                     */
+                    spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
+                }
+            } else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
+                memset(&spec[pos], 0, num_specs*sizeof(float));
+            }
+        }
+    }
+
+    return 0;
+}
+
+
+void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
+{
+    float   temp[256];
+    float   iqmf_temp[512 + 46];
+
+    /* combine low and middle bands */
+    atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
+
+    /* delay the signal of the high band by 23 samples */
+    memcpy( su->last_qmf_delay,     &su->last_qmf_delay[256], sizeof(float)*23);
+    memcpy(&su->last_qmf_delay[23], q->bands[2],       sizeof(float)*256);
+
+    /* combine (low + middle) and high bands */
+    atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
+}
+
+
+static int atrac1_decode_frame(AVCodecContext *avctx,
+            void *data, int *data_size,
+            AVPacket *avpkt)
+{
+    const uint8_t *buf = avpkt->data;
+    int buf_size = avpkt->size;
+    AT1Ctx *q = avctx->priv_data;
+    int ch, ret, i;
+    GetBitContext gb;
+    float* samples = data;
+
+
+    if (buf_size < 212 * q->channels) {
+        av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n");
+        return -1;
+    }
+
+    for (ch=0 ; ch<q->channels ; ch++) {
+        AT1SUCtx* su = &q->SUs[ch];
+
+        init_get_bits(&gb, &buf[212*ch], 212*8);
+
+        /* parse block_size_mode, 1st byte */
+        ret = at1_parse_block_size_mode(&gb, su->log2_block_count);
+        if (ret < 0)
+            return ret;
+
+        ret = at1_unpack_dequant(&gb, su, q->spec);
+        if (ret < 0)
+            return ret;
+
+        ret = at1_imdct_block(su, q);
+        if (ret < 0)
+            return ret;
+        at1_subband_synthesis(q, su, q->out_samples[ch]);
+    }
+
+    /* round, convert to 16bit and interleave */
+    if (q->channels == 1) {
+        /* mono */
+        q->dsp.vector_clipf(samples, q->out_samples[0], -32700./(1<<15), 32700./(1<<15), AT1_SU_SAMPLES);
+    } else {
+        /* stereo */
+        for (i = 0; i < AT1_SU_SAMPLES; i++) {
+            samples[i*2]   = av_clipf(q->out_samples[0][i], -32700./(1<<15), 32700./(1<<15));
+            samples[i*2+1] = av_clipf(q->out_samples[1][i], -32700./(1<<15), 32700./(1<<15));
+        }
+    }
+
+    *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
+    return avctx->block_align;
+}
+
+
+static av_cold void init_mdct_windows(void)
+{
+    int i;
+
+    /** The mid and long windows uses the same sine window splitted
+     *  in the middle and wrapped into zero/one regions as follows:
+     *
+     *                   region of "ones"
+     *               -------------
+     *              /
+     *             / 1st half
+     *            / of the sine
+     *           /  window
+     * ---------/
+     * zero region
+     *
+     * The mid and short windows are subsets of the long window.
+     */
+
+    /* Build "zero" region */
+    memset(long_window, 0, sizeof(long_window));
+    /* Build sine window region */
+    short_window = &long_window[112];
+    ff_sine_window_init(short_window,32);
+    /* Build "ones" region */
+    for (i = 0; i < 112; i++)
+        long_window[144 + i] = 1.0f;
+    /* Save the mid window subset start */
+    mid_window = &long_window[64];
+
+    /* Prepare the window table */
+    window_per_band[0] = mid_window;
+    window_per_band[1] = mid_window;
+    window_per_band[2] = long_window;
+}
+
+static av_cold int atrac1_decode_init(AVCodecContext *avctx)
+{
+    AT1Ctx *q = avctx->priv_data;
+
+    avctx->sample_fmt = SAMPLE_FMT_FLT;
+
+    q->channels = avctx->channels;
+
+    /* Init the mdct transforms */
+    ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1<<15));
+    ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1<<15));
+    ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1<<15));
+    init_mdct_windows();
+
+    atrac_generate_tables();
+
+    dsputil_init(&q->dsp, avctx);
+
+    q->bands[0] = q->low;
+    q->bands[1] = q->mid;
+    q->bands[2] = q->high;
+
+    /* Prepare the mdct overlap buffers */
+    q->SUs[0].spectrum[0] = q->SUs[0].spec1;
+    q->SUs[0].spectrum[1] = q->SUs[0].spec2;
+    q->SUs[1].spectrum[0] = q->SUs[1].spec1;
+    q->SUs[1].spectrum[1] = q->SUs[1].spec2;
+
+    return 0;
+}
+
+AVCodec atrac1_decoder = {
+    .name = "atrac1",
+    .type = CODEC_TYPE_AUDIO,
+    .id = CODEC_ID_ATRAC1,
+    .priv_data_size = sizeof(AT1Ctx),
+    .init = atrac1_decode_init,
+    .close = NULL,
+    .decode = atrac1_decode_frame,
+    .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
+};